1. Field of the Invention
The present invention relates to a system and method for enhancing voice channel transmission in a voice over Internet Protocol (IP), and more specifically, to a method for providing an efficient header transport so as to increase the bandwidth available for data payload transport.
2. Background of the Prior art
In the rapidly changing, deregulated telecommunications marketplace, Internet Service Providers (ISPs) can capitalize on the growing opportunities by introducing value-added services to customers seeking more reliable and economical telecommunications services. The tremendous growth in network infrastructure and corresponding growth in Internet Protocol (IP) traffic has resulted in the use of IP as a platform for new telecommunications services that take advantage of voice and data convergence. Voice over IP (VOIP) technology is one such telecommunications service.
Prior art voice and data convergence (i.e., carrying voice and data within one copper line) has resulted in prior art telephone service adapting from a circuit network to a packet network environment. More specifically, various industry groups are approving protocols and standards (e.g., ITU-T H.323, IETF SIP, H.248, etc.) for VOIP. As a result, prior art VOIP services are available for customers to use VOIP via Internet phone service from a personal computer (PC) or an Ethernet IP phone (e.g., Nortel i2004 telephone, Cisco 7960 telephone), including high speed Internet communications via a high speed access network (e.g., xDSL, Cable modem or WLL).
However, the prior art voice and data convergence provides limited opportunities for CLECs (Competitive Local Exchange Carriers) and ILECs (Incumbent Local Exchange Carriers) to provide VOIP, due to infrastructure limitations. For example, CLECs only have data lines, and would prefer to only transmit data whereas ILECs only have voice lines, and would prefer to provide voice transport due to the high cost of adding data lines. Further, voice and data convergence requires infrastructure for managing multiple telephone lines for customers to use in a home or a small office/home office. In the prior art, these multiple lines are managed at an IAD (Integrated Access Device), which is a terminal where voice and data converge, such that the customer has concurrent data and voice service.
In addition to infrastructure requirements, there is a lack of standardization in VOIP. For example, while the DSL Forum approved the prior art technology and protocol to transmit voice over DSL (called VoDSL) in AAL5, also known as TR-036, in August 2000, the Asynchronous Transfer Mode (ATM) Forum has not yet approved AF-VMOA-0145.000, which uses AAL2. Thus, there is a prior art disadvantage in that no AAL2 standard relating to VOIP over DSL has been approved.
FIG. 1 illustrates a prior art VOIP system 1 configured for phone users to access the Internet via an ADSL (Asymmetric Digital Subscriber Line) network, including a home network having user data devices 3a . . . 3n and a modem 5, an access network having a Digital Subscriber Line Access Multiplier (DSLAM) 11 and a Registration, Administration and Status (RAS) server 13, and a backbone network including the Internet 17 and a VOIP server 19. The user data devices 3a . . . 3n, can be PCs, Internet phones, conventional phones, or any combination thereof. The ADSL Modem (i.e., ATU-R) 5 receives voice data from the user device 3a via a prior art external interface (e.g., 10BaseT Ethernet, USB, HomePNA) 7, and transmits this voice data to the DSLAM 11 via a copper wire 9. The voice data is then transmitted to the RAS server 13 via an STM-1 connection 15, and eventually to the Internet 17. A VOIP server 19 connected to the Internet 17 provides a prior art interface (e.g., web site) between the receiving user and the transmitting user for routing the voice data to a RAS 13, DSLAM 11 and modem 5 on the receiving side.
As illustrated in FIG. 1, in a prior art method, a user can connect to an Internet service, which requires turning on the modem 5, which in turn attempts to connect to the DSLAM 11. After navigating the Internet 17 for a particular site (e.g., www.dialpad.com), the user, using the prior art RAS 13 via a PPP server, connects to the VOIP server 19, which attempts to contact the receiver of the user's message and make a link. Once the user hits the “send” button on the Internet site to send the phone number, the VOIP server searches a database (i.e., directory) based on the phone number supplied by the user, and attempts to link the user to a destination.
FIG. 2 illustrates the prior art VOIP protocol stack, wherein ITU-T H.323 is used as VOIP protocol to transmit voice over an IP network. An H.323/H.323 gateway forms a multimedia/voice standard for communication between endpoints, and can be used for the VOIP protocol, and covers multimedia. The transport layer always includes RTP/UDP protocol information, and the network layer always includes IP header information.
FIG. 3 illustrates a telephone call placed according to a H.323 call procedure. In phase A, H.225.0 is used for call setup, whereas phase B, phase C and phase E use H.245, and phase E also uses H.225.0. After establishment of communication in phase C, the information flow occurs in phase D, which includes media stream and media stream control flows. It is noted that phase D is not in H.245. Phase D is the heaviest and most intense phase for data processing, whereas the other phases do not have as heavy a load. Because the phases other than phase D are each performed as a single shot, those phases do not load as heavily into the ADSL link. It is expected that RTP will be adapted as the protocol for transmitting voice signals. Thus, phase D using the RTP phase will most heavily load the network.
In the prior art VOIP system 1, the link speed can vary according to the company providing the service. The prior art VOIP system 1 enhances the bandwidth of the prior art copper line, used for approximately 100 years, from 4 KHz bandwidth to 8.1M/0.8 Mbps (downstream/upstream), which is the fastest possible prior art link speed. When the prior art ADSL system 1 is used for symmetric (e.g., voice) data transport, the data transfer rate is 384/384 Kbps, 768/768 Kbps, and differs based on distance. While data service requires a greater download speed than upload speed, voice service requires a substantially similar download and upload speed due to the continuous nature of voice conversation. Thus, the limit of voice service provided by the prior art IAD with ADSL depends on upload speed, which is lower than the download speed.
FIG. 4 illustrates a prior art H.323 data transmission architecture 21. When adapting the prior art H.323 protocol to VOIP, a voice signal is transformed to a digital signal having RTP/UDP/IP headers in the IAD. As illustrated in FIG. 4, a 20-byte IP header 23, followed by an 8-byte UDP header 25 and a 12-byte RTP header 27, are sent in each cell prior to a 20-byte payload 29. Further, the voice signal is sent by a G.726-32 voice codec, and the voice codec generates a 20-byte voice payload at every 5 ms (20-byte/5 ms=4000-byte/1 s=32 Kbps), such that the 20-byte voice payload is inserted into the IP packet. Thus, the IP packet has a size of 60-bytes. However, there is only a 20-byte payload on the 60-byte packet. Thus, the prior art is inefficient in that only 20 of 60 possible bytes, or 33%, are used for payload.
Further, as illustrated in FIG. 4, two AAL5 cells 31, 37 are required to send the voice data. For example, a first 5-byte ATM header 33 is followed by a 48-byte payload 35, and then a second 5-byte ATM header 39 is followed by a 12-byte payload 41, a pad 43 and a tailer 45. Thus, the prior art AAL5 cell has a disadvantage in that the data payload cannot be transported in a single packet due to the high overhead requirements.
FIG. 5 illustrates the prior art VOIP method. Voice packets are transmitted via the cells illustrated in FIG. 4. For example, a phone 3a is attached, via analog line 2, to an IAD 22. The IAD 22 is attached to the DSLAM 11 via an ADSL line, as described above, and the DSLAM 11 is connected to the RAS 13 via an OC-3 or STM-1 line 6, as also described above. The RAS 13 then transmits IP packets to the Internet 17.
In the prior art method of FIG. 5, IP and RTP/UDP protocol information are initially positioned in each AAL5 cell when the data is set up for IP transport at the IAD 22. The AAL5 cell is then transported from the IAD 22 to the RAS 13 via AAL5. The RAS 13 removes AAL5 information and transmits the IP datagram to the Internet 17. The link and physical information is removed prior to the RAS 13 receiving the packet. Thus, the extra overhead of sending the IP header and the RTP/UDP header from the IAD 22 to the RAS 13 have the disadvantage of creating the above-described inefficiencies for the prior art AAL5 transfer method. Accordingly, upload speed is reduced, and overall operational speed is limited, because as noted above, the upload and download speeds must be substantially identical for effective voice communication. Also, fewer concurrent channels (e.g., calls) can be maintained at an adequate quality.
Table 1 shows performance for various prior art code (i.e., code-decode) formats, wherein a low MOS corresponds to a low conversation quality:
AvailableCodecMOSRatePayload SizeNeeded cellscallsG.7114.3 64 Kbps40 + 40 bytes2-cell/5 ms5G.7263.7 32 Kbps20 + 40 bytes2-cell/5 ms5G.7284.0 16 Kbps10 + 40 bytes2-cell/5 ms5G.7293.8  8 Kbps10 + 40 bytes2-cell/10 ms10G.723.13.56.3 Kbps20 + 40 bytes2-cell/30 ms30
The voice transfer rate for loading cells into the ADSL link of the AAL5 prior art example is about 153.6 Kbps (2-cell/5 ms=400-cell/s=400*48*8 bps=153.6 Kbps). Thus, the IAD only supports only 5 concurrent calls at an adequate quality (e.g., MOS greater than or equal to 4.0) because of the 800 Kbps upper limit on upload speed. However, this data rate is calculated for voice service only. If the IAD also supports data service, the number of available concurrent calls will be reduced even further. When G.711-64 is used as codec and the frame interval is 5 ms, a 40-byte voice payload results in 80-byte RTP packet due to overhead. Accordingly, 2 AAL5 cells are required to transmit the RTP packet over AAL5.
However, the prior art has various problems and disadvantages. For example, because the upload speed is so much more limited than the downlink speed, there is a shortage of uplink bandwidth due to the prior art method of data transfer. Further, because the UDP header, RTP header and IP header are included in every packet transmitted to the DSLAM 11, VOIP data transport is extremely inefficient (on the order of 67% overhead per data packet), and the AAL5 model for data transport requires more than one packet to transport data. Thus, fewer concurrent calls can be maintained, and the Quality of Service (QoS) cannot be guaranteed. Further, the prior art use of the H.323 protocol (e.g., www.dialpad.com) is complex and difficult to implement (e.g., too narrow bandwidth for video telephone conferencing), and only one payload of voice data can be transmitted per cell.